UnrealXR/raylib
2
0
Fork 0
Code Issues Pull requests Projects Releases Packages Wiki Activity Actions

WARNING: BREAKING: Use frameCount on audio

Browse source 
This is a big change for optimization and a more professional understanding of audio. Instead of dealing with samples, now we deal with frames, like miniaudio does, so, avoiding continuous conversions from samples to frames.
This commit is contained in:
Ray 2021-08-16 11:06:31 +02:00
parent e203fb58c6
commit 1b4c58b66f
3 changed files with 105 additions and 118 deletions
Download patch file Download diff file Expand all files Collapse all files

160
src/raudio.c
View file

@ -728,11 +728,11 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
if (success)
{
wave.sampleCount = (unsigned int)wav.totalPCMFrameCount*wav.channels;
wave.frameCount = (unsigned int)wav.totalPCMFrameCount;
wave.sampleRate = wav.sampleRate;
wave.sampleSize = 16;
wave.channels = wav.channels;
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
// NOTE: We are forcing conversion to 16bit sample size on reading
drwav_read_pcm_frames_s16(&wav, wav.totalPCMFrameCount, wave.data);
@ -754,11 +754,11 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
wave.sampleRate = info.sample_rate;
wave.sampleSize = 16; // By default, ogg data is 16 bit per sample (short)
wave.channels = info.channels;
wave.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData)*info.channels; // Independent by channel
wave.data = (short *)RL_MALLOC(wave.sampleCount*sizeof(short));
wave.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples(oggData); // NOTE: It returns frames!
wave.data = (short *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(short));
// NOTE: Get the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.sampleCount);
// NOTE: Get the number of samples to process (be careful! we ask for number of shorts, not bytes!)
stb_vorbis_get_samples_short_interleaved(oggData, info.channels, (short *)wave.data, wave.frameCount*wave.channels);
stb_vorbis_close(oggData);
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load OGG data");
@ -773,7 +773,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
wave.data = drflac_open_memory_and_read_pcm_frames_s16(fileData, dataSize, &wave.channels, &wave.sampleRate, &totalFrameCount, NULL);
wave.sampleSize = 16;
if (wave.data != NULL) wave.sampleCount = (unsigned int)totalFrameCount*wave.channels;
if (wave.data != NULL) wave.frameCount = (unsigned int)totalFrameCount;
else TRACELOG(LOG_WARNING, "WAVE: Failed to load FLAC data");
}
#endif
@ -791,7 +791,7 @@ Wave LoadWaveFromMemory(const char *fileType, const unsigned char *fileData, int
{
wave.channels = config.channels;
wave.sampleRate = config.sampleRate;
wave.sampleCount = (int)totalFrameCount*wave.channels;
wave.frameCount = (int)totalFrameCount;
}
else TRACELOG(LOG_WARNING, "WAVE: Failed to load MP3 data");
@ -835,7 +835,7 @@ Sound LoadSoundFromWave(Wave wave)
// First option has been selected, format conversion is done on the loading stage.
// The downside is that it uses more memory if the original sound is u8 or s16.
ma_format formatIn = ((wave.sampleSize == 8)? ma_format_u8 : ((wave.sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave.sampleCount/wave.channels;
ma_uint32 frameCountIn = wave.frameCount;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, NULL, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed to get frame count for format conversion");
@ -850,7 +850,7 @@ Sound LoadSoundFromWave(Wave wave)
frameCount = (ma_uint32)ma_convert_frames(audioBuffer->data, frameCount, AUDIO_DEVICE_FORMAT, AUDIO_DEVICE_CHANNELS, AUDIO.System.device.sampleRate, wave.data, frameCountIn, formatIn, wave.channels, wave.sampleRate);
if (frameCount == 0) TRACELOG(LOG_WARNING, "SOUND: Failed format conversion");
sound.sampleCount = frameCount*AUDIO_DEVICE_CHANNELS;
sound.frameCount = frameCount;
sound.stream.sampleRate = AUDIO.System.device.sampleRate;
sound.stream.sampleSize = 32;
sound.stream.channels = AUDIO_DEVICE_CHANNELS;
@ -908,7 +908,7 @@ bool ExportWave(Wave wave, const char *fileName)
void *fileData = NULL;
size_t fileDataSize = 0;
success = drwav_init_memory_write(&wav, &fileData, &fileDataSize, &format, NULL);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.sampleCount/wave.channels, wave.data);
if (success) success = (int)drwav_write_pcm_frames(&wav, wave.frameCount, wave.data);
drwav_result result = drwav_uninit(&wav);
if (result == DRWAV_SUCCESS) success = SaveFileData(fileName, (unsigned char *)fileData, (unsigned int)fileDataSize);
@ -920,7 +920,7 @@ bool ExportWave(Wave wave, const char *fileName)
{
// Export raw sample data (without header)
// NOTE: It's up to the user to track wave parameters
success = SaveFileData(fileName, wave.data, wave.sampleCount*wave.sampleSize/8);
success = SaveFileData(fileName, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
}
if (success) TRACELOG(LOG_INFO, "FILEIO: [%s] Wave data exported successfully", fileName);
@ -938,7 +938,7 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
#define TEXT_BYTES_PER_LINE 20
#endif
int waveDataSize = wave.sampleCount*wave.channels*wave.sampleSize/8;
int waveDataSize = wave.frameCount*wave.channels*wave.sampleSize/8;
// NOTE: Text data buffer size is estimated considering wave data size in bytes
// and requiring 6 char bytes for every byte: "0x00, "
@ -966,7 +966,8 @@ bool ExportWaveAsCode(Wave wave, const char *fileName)
#endif
bytesCount += sprintf(txtData + bytesCount, "// Wave data information\n");
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.sampleCount);
bytesCount += sprintf(txtData + bytesCount, "#define %s_FRAME_COUNT %u\n", varFileName, wave.frameCount);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_COUNT %u\n", varFileName, wave.frameCount*wave.channels);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_RATE %u\n", varFileName, wave.sampleRate);
bytesCount += sprintf(txtData + bytesCount, "#define %s_SAMPLE_SIZE %u\n", varFileName, wave.sampleSize);
bytesCount += sprintf(txtData + bytesCount, "#define %s_CHANNELS %u\n\n", varFileName, wave.channels);
@ -1111,7 +1112,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
ma_format formatIn = ((wave->sampleSize == 8)? ma_format_u8 : ((wave->sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_format formatOut = ((sampleSize == 8)? ma_format_u8 : ((sampleSize == 16)? ma_format_s16 : ma_format_f32));
ma_uint32 frameCountIn = wave->sampleCount/wave->channels;
ma_uint32 frameCountIn = wave->frameCount;
ma_uint32 frameCount = (ma_uint32)ma_convert_frames(NULL, 0, formatOut, channels, sampleRate, NULL, frameCountIn, formatIn, wave->channels, wave->sampleRate);
if (frameCount == 0)
@ -1129,7 +1130,7 @@ void WaveFormat(Wave *wave, int sampleRate, int sampleSize, int channels)
return;
}
wave->sampleCount = frameCount*channels;
wave->frameCount = frameCount;
wave->sampleSize = sampleSize;
wave->sampleRate = sampleRate;
wave->channels = channels;
@ -1142,14 +1143,14 @@ Wave WaveCopy(Wave wave)
{
Wave newWave = { 0 };
newWave.data = RL_MALLOC(wave.sampleCount*wave.sampleSize/8);
newWave.data = RL_MALLOC(wave.frameCount*wave.channels*wave.sampleSize/8);
if (newWave.data != NULL)
{
// NOTE: Size must be provided in bytes
memcpy(newWave.data, wave.data, wave.sampleCount*wave.sampleSize/8);
memcpy(newWave.data, wave.data, wave.frameCount*wave.channels*wave.sampleSize/8);
newWave.sampleCount = wave.sampleCount;
newWave.frameCount = wave.frameCount;
newWave.sampleRate = wave.sampleRate;
newWave.sampleSize = wave.sampleSize;
newWave.channels = wave.channels;
@ -1163,7 +1164,7 @@ Wave WaveCopy(Wave wave)
void WaveCrop(Wave *wave, int initSample, int finalSample)
{
if ((initSample >= 0) && (initSample < finalSample) &&
(finalSample > 0) && ((unsigned int)finalSample < wave->sampleCount))
(finalSample > 0) && ((unsigned int)finalSample < (wave->frameCount*wave->channels)))
{
int sampleCount = finalSample - initSample;
@ -1182,11 +1183,11 @@ void WaveCrop(Wave *wave, int initSample, int finalSample)
// NOTE 2: Sample data allocated should be freed with UnloadWaveSamples()
float *LoadWaveSamples(Wave wave)
{
float *samples = (float *)RL_MALLOC(wave.sampleCount*sizeof(float));
float *samples = (float *)RL_MALLOC(wave.frameCount*wave.channels*sizeof(float));
// NOTE: sampleCount is the total number of interlaced samples (including channels)
for (unsigned int i = 0; i < wave.sampleCount; i++)
for (unsigned int i = 0; i < wave.frameCount*wave.channels; i++)
{
if (wave.sampleSize == 8) samples[i] = (float)(((unsigned char *)wave.data)[i] - 127)/256.0f;
else if (wave.sampleSize == 16) samples[i] = (float)(((short *)wave.data)[i])/32767.0f;
@ -1228,7 +1229,7 @@ Music LoadMusicStream(const char *fileName)
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1249,7 +1250,7 @@ Music LoadMusicStream(const char *fileName)
music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1266,7 +1267,7 @@ Music LoadMusicStream(const char *fileName)
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1284,7 +1285,7 @@ Music LoadMusicStream(const char *fileName)
if (result > 0)
{
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1309,9 +1310,9 @@ Music LoadMusicStream(const char *fileName)
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
musicLoaded = true;
}
}
@ -1330,7 +1331,7 @@ Music LoadMusicStream(const char *fileName)
{
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, AUDIO_DEVICE_CHANNELS);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1367,10 +1368,10 @@ Music LoadMusicStream(const char *fileName)
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: [%s] Music file loaded successfully", fileName);
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
@ -1402,7 +1403,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
if (ctxWav->bitsPerSample == 24) sampleSize = 16; // Forcing conversion to s16 on UpdateMusicStream()
music.stream = LoadAudioStream(ctxWav->sampleRate, sampleSize, ctxWav->channels);
music.sampleCount = (unsigned int)ctxWav->totalPCMFrameCount*ctxWav->channels;
music.frameCount = (unsigned int)ctxWav->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1419,7 +1420,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
drflac *ctxFlac = (drflac *)music.ctxData;
music.stream = LoadAudioStream(ctxFlac->sampleRate, ctxFlac->bitsPerSample, ctxFlac->channels);
music.sampleCount = (unsigned int)ctxFlac->totalPCMFrameCount*ctxFlac->channels;
music.frameCount = (unsigned int)ctxFlac->totalPCMFrameCount;
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1437,7 +1438,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
if (success)
{
music.stream = LoadAudioStream(ctxMp3->sampleRate, 32, ctxMp3->channels);
music.sampleCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3)*ctxMp3->channels;
music.frameCount = (unsigned int)drmp3_get_pcm_frame_count(ctxMp3);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1459,7 +1460,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
music.stream = LoadAudioStream(info.sample_rate, 16, info.channels);
// WARNING: It seems this function returns length in frames, not samples, so we multiply by channels
music.sampleCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData)*info.channels;
music.frameCount = (unsigned int)stb_vorbis_stream_length_in_samples((stb_vorbis *)music.ctxData);
music.looping = true; // Looping enabled by default
musicLoaded = true;
}
@ -1483,7 +1484,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
// NOTE: Only stereo is supported for XM
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, bits, 2);
music.sampleCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm)*2; // 2 channels
music.frameCount = (unsigned int)jar_xm_get_remaining_samples(ctxXm); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
jar_xm_reset(ctxXm); // make sure we start at the beginning of the song
@ -1521,7 +1522,7 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
// NOTE: Only stereo is supported for MOD
music.stream = LoadAudioStream(AUDIO.System.device.sampleRate, 16, 2);
music.sampleCount = (unsigned int)jar_mod_max_samples(ctxMod)*2; // 2 channels
music.frameCount = (unsigned int)jar_mod_max_samples(ctxMod); // NOTE: Always 2 channels (stereo)
music.looping = true; // Looping enabled by default
musicLoaded = true;
@ -1561,10 +1562,10 @@ Music LoadMusicStreamFromMemory(const char *fileType, unsigned char *data, int d
{
// Show some music stream info
TRACELOG(LOG_INFO, "FILEIO: Music data loaded successfully");
TRACELOG(LOG_INFO, " > Total samples: %i", music.sampleCount);
TRACELOG(LOG_INFO, " > Sample rate: %i Hz", music.stream.sampleRate);
TRACELOG(LOG_INFO, " > Sample size: %i bits", music.stream.sampleSize);
TRACELOG(LOG_INFO, " > Channels: %i (%s)", music.stream.channels, (music.stream.channels == 1)? "Mono" : (music.stream.channels == 2)? "Stereo" : "Multi");
TRACELOG(LOG_INFO, " > Total frames: %i", music.frameCount);
}
return music;
@ -1660,29 +1661,22 @@ void UpdateMusicStream(Music music)
{
if (music.stream.buffer == NULL) return;
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM) jar_xm_set_max_loop_count(music.ctxData, music.looping ? 0 : 1);
#endif
bool streamEnding = false;
unsigned int subBufferSizeInFrames = music.stream.buffer->sizeInFrames/2;
// NOTE: Using dynamic allocation because it could require more than 16KB
void *pcm = RL_CALLOC(subBufferSizeInFrames*music.stream.channels*music.stream.sampleSize/8, 1);
int samplesCount = 0; // Total size of data streamed in L+R samples for xm floats, individual L or R for ogg shorts
int frameCountToStream = 0; // Total size of data in frames to be streamed
// TODO: Get the sampleLeft using framesProcessed... but first, get total frames processed correctly...
// TODO: Get the framesLeft using framesProcessed... but first, get total frames processed correctly...
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
int sampleLeft = music.sampleCount - (music.stream.buffer->framesProcessed*music.stream.channels);
if (music.ctxType == MUSIC_MODULE_XM && music.looping) sampleLeft = subBufferSizeInFrames*4;
int framesLeft = music.frameCount - music.stream.buffer->framesProcessed;
while (IsAudioStreamProcessed(music.stream))
{
if ((sampleLeft/music.stream.channels) >= subBufferSizeInFrames) samplesCount = subBufferSizeInFrames*music.stream.channels;
else samplesCount = sampleLeft;
if (framesLeft >= subBufferSizeInFrames) frameCountToStream = subBufferSizeInFrames;
else frameCountToStream = framesLeft;
switch (music.ctxType)
{
@ -1690,8 +1684,8 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_WAV:
{
// NOTE: Returns the number of samples to process (not required)
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, samplesCount/music.stream.channels, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
if (music.stream.sampleSize == 16) drwav_read_pcm_frames_s16((drwav *)music.ctxData, frameCountToStream, (short *)pcm);
else if (music.stream.sampleSize == 32) drwav_read_pcm_frames_f32((drwav *)music.ctxData, frameCountToStream, (float *)pcm);
} break;
#endif
@ -1699,7 +1693,7 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_OGG:
{
// NOTE: Returns the number of samples to process (be careful! we ask for number of shorts!)
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, samplesCount);
stb_vorbis_get_samples_short_interleaved((stb_vorbis *)music.ctxData, music.stream.channels, (short *)pcm, frameCountToStream*music.stream.channels);
} break;
#endif
@ -1707,15 +1701,14 @@ void UpdateMusicStream(Music music)
case MUSIC_AUDIO_FLAC:
{
// NOTE: Returns the number of samples to process (not required)
drflac_read_pcm_frames_s16((drflac *)music.ctxData, samplesCount, (short *)pcm);
drflac_read_pcm_frames_s16((drflac *)music.ctxData, frameCountToStream*music.stream.channels, (short *)pcm);
} break;
#endif
#if defined(SUPPORT_FILEFORMAT_MP3)
case MUSIC_AUDIO_MP3:
{
// NOTE: samplesCount, actually refers to framesCount and returns the number of frames processed
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, samplesCount/music.stream.channels, (float *)pcm);
drmp3_read_pcm_frames_f32((drmp3 *)music.ctxData, frameCountToStream, (float *)pcm);
} break;
#endif
@ -1723,9 +1716,9 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_XM:
{
// NOTE: Internally we consider 2 channels generation, so samplesCount/2
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)pcm, samplesCount/2);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, samplesCount/2);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, samplesCount/2);
if (AUDIO_DEVICE_FORMAT == ma_format_f32) jar_xm_generate_samples((jar_xm_context_t *)music.ctxData, (float *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_s16) jar_xm_generate_samples_16bit((jar_xm_context_t *)music.ctxData, (short *)pcm, frameCountToStream);
else if (AUDIO_DEVICE_FORMAT == ma_format_u8) jar_xm_generate_samples_8bit((jar_xm_context_t *)music.ctxData, (char *)pcm, frameCountToStream);
} break;
#endif
@ -1733,22 +1726,17 @@ void UpdateMusicStream(Music music)
case MUSIC_MODULE_MOD:
{
// NOTE: 3rd parameter (nbsample) specify the number of stereo 16bits samples you want, so sampleCount/2
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, samplesCount/2, 0);
jar_mod_fillbuffer((jar_mod_context_t *)music.ctxData, (short *)pcm, frameCountToStream, 0);
} break;
#endif
default: break;
}
UpdateAudioStream(music.stream, pcm, samplesCount);
UpdateAudioStream(music.stream, pcm, frameCountToStream);
framesLeft -= frameCountToStream;
if ((music.ctxType == MUSIC_MODULE_XM) || music.ctxType == MUSIC_MODULE_MOD)
{
if (samplesCount > 1) sampleLeft -= samplesCount/2;
else sampleLeft -= samplesCount;
}
else sampleLeft -= samplesCount;
if (sampleLeft <= 0)
if (framesLeft <= 0)
{
streamEnding = true;
break;
@ -1795,7 +1783,7 @@ float GetMusicTimeLength(Music music)
{
float totalSeconds = 0.0f;
totalSeconds = (float)music.sampleCount/(music.stream.sampleRate*music.stream.channels);
totalSeconds = (float)music.frameCount/music.stream.sampleRate;
return totalSeconds;
}
@ -1803,22 +1791,24 @@ float GetMusicTimeLength(Music music)
// Get current music time played (in seconds)
float GetMusicTimePlayed(Music music)
{
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t samples = 0;
jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &samples);
samples = samples % (music.sampleCount);
return (float)(samples)/(music.stream.sampleRate*music.stream.channels);
}
#endif
float secondsPlayed = 0.0f;
if (music.stream.buffer != NULL)
{
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int samplesPlayed = music.stream.buffer->framesProcessed*music.stream.channels;
secondsPlayed = (float)samplesPlayed/(music.stream.sampleRate*music.stream.channels);
#if defined(SUPPORT_FILEFORMAT_XM)
if (music.ctxType == MUSIC_MODULE_XM)
{
uint64_t framesPlayed = 0;
jar_xm_get_position(music.ctxData, NULL, NULL, NULL, &framesPlayed);
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
else
#endif
{
//ma_uint32 frameSizeInBytes = ma_get_bytes_per_sample(music.stream.buffer->dsp.formatConverterIn.config.formatIn)*music.stream.buffer->dsp.formatConverterIn.config.channels;
unsigned int framesPlayed = music.stream.buffer->framesProcessed;
secondsPlayed = (float)framesPlayed/music.stream.sampleRate;
}
}
return secondsPlayed;
@ -1867,7 +1857,7 @@ void UnloadAudioStream(AudioStream stream)
// Update audio stream buffers with data
// NOTE 1: Only updates one buffer of the stream source: unqueue -> update -> queue
// NOTE 2: To unqueue a buffer it needs to be processed: IsAudioStreamProcessed()
void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
void UpdateAudioStream(AudioStream stream, const void *data, int frameCount)
{
if (stream.buffer != NULL)
{
@ -1896,11 +1886,11 @@ void UpdateAudioStream(AudioStream stream, const void *data, int samplesCount)
// Does this API expect a whole buffer to be updated in one go?
// Assuming so, but if not will need to change this logic.
if (subBufferSizeInFrames >= (ma_uint32)samplesCount/stream.channels)
if (subBufferSizeInFrames >= (ma_uint32)frameCount)
{
ma_uint32 framesToWrite = subBufferSizeInFrames;
if (framesToWrite > ((ma_uint32)samplesCount/stream.channels)) framesToWrite = (ma_uint32)samplesCount/stream.channels;
if (framesToWrite > (ma_uint32)frameCount) framesToWrite = (ma_uint32)frameCount;
ma_uint32 bytesToWrite = framesToWrite*stream.channels*(stream.sampleSize/8);
memcpy(subBuffer, data, bytesToWrite);